CA2908891A1 - A computer-implemented method for reducing crosstalk in a computer-based audiometer - Google Patents
A computer-implemented method for reducing crosstalk in a computer-based audiometer Download PDFInfo
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R3/00—Circuits for transducers, loudspeakers or microphones
- H04R3/12—Circuits for transducers, loudspeakers or microphones for distributing signals to two or more loudspeakers
- H04R3/14—Cross-over networks
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- A—HUMAN NECESSITIES
- A61—MEDICAL OR VETERINARY SCIENCE; HYGIENE
- A61B—DIAGNOSIS; SURGERY; IDENTIFICATION
- A61B5/00—Measuring for diagnostic purposes; Identification of persons
- A61B5/12—Audiometering
- A61B5/121—Audiometering evaluating hearing capacity
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
- G10K11/00—Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
- G10K11/16—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
- G10K11/175—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
- G10K11/178—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase
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- H—ELECTRICITY
- H03—ELECTRONIC CIRCUITRY
- H03G—CONTROL OF AMPLIFICATION
- H03G3/00—Gain control in amplifiers or frequency changers
- H03G3/20—Automatic control
- H03G3/30—Automatic control in amplifiers having semiconductor devices
- H03G3/32—Automatic control in amplifiers having semiconductor devices the control being dependent upon ambient noise level or sound level
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- H—ELECTRICITY
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- H04B—TRANSMISSION
- H04B3/00—Line transmission systems
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- H04R5/00—Stereophonic arrangements
- H04R5/04—Circuit arrangements, e.g. for selective connection of amplifier inputs/outputs to loudspeakers, for loudspeaker detection, or for adaptation of settings to personal preferences or hearing impairments
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- A—HUMAN NECESSITIES
- A61—MEDICAL OR VETERINARY SCIENCE; HYGIENE
- A61B—DIAGNOSIS; SURGERY; IDENTIFICATION
- A61B2505/00—Evaluating, monitoring or diagnosing in the context of a particular type of medical care
- A61B2505/09—Rehabilitation or training
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- A61—MEDICAL OR VETERINARY SCIENCE; HYGIENE
- A61B—DIAGNOSIS; SURGERY; IDENTIFICATION
- A61B2560/00—Constructional details of operational features of apparatus; Accessories for medical measuring apparatus
- A61B2560/02—Operational features
- A61B2560/0223—Operational features of calibration, e.g. protocols for calibrating sensors
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- A61B2560/04—Constructional details of apparatus
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- A61—MEDICAL OR VETERINARY SCIENCE; HYGIENE
- A61B—DIAGNOSIS; SURGERY; IDENTIFICATION
- A61B5/00—Measuring for diagnostic purposes; Identification of persons
- A61B5/68—Arrangements of detecting, measuring or recording means, e.g. sensors, in relation to patient
- A61B5/6801—Arrangements of detecting, measuring or recording means, e.g. sensors, in relation to patient specially adapted to be attached to or worn on the body surface
- A61B5/6813—Specially adapted to be attached to a specific body part
- A61B5/6814—Head
- A61B5/6815—Ear
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Abstract
A computer-implemented method for reducing undesired crosstalk signals on an inactive channel of a device comprising the steps of: (i) determining system volume levels and associated signal amplitudes required to achieve a range of desired audio output attenuation levels on an active channel of the device; (ii) determining a crosstalk compensation signal comprising a signal amplitude and associated phase shift required to reduce undesired crosstalk on the inactive channel of the device for each desired audio output attention level in the range of desired audio output attenuation levels; and (iii) generating a desired audio output attenuation level on the active channel of the device by generating a signal at the determined system volume level and associated signal amplitude required to achieve said desired audio output attenuation level, and generating a contemporaneous crosstalk compensation signal on the inactive channel of the device by generating a signal at the determined signal amplitude and associated phase shift required to reduce the undesired crosstalk on the inactive channel.
Description
A COMPUTER-IMPLEMENTED METHOD FOR REDUCING CROSSTALK IN A
COMPUTER-BASED AUDIOMETER
A crosstalk reduction system and method is disclosed for use in a computer-based audiometer. Crosstalk which couples between channels is reduced through use of a software module which modifies digital samples being sent to a digital-to-analog converter.
Crosstalk, the unwanted transmission of signals between communication channels, in the context of hearing testing means that an unwanted audible signal is present on one earphone when the test signal is being directed to the earphone of the opposite side.
Amplifier Jack/Plug Headphones Left IL '4 M1 g r _________________________________________________________________ RI
Rig < <
Right Ground Audible crosstalk occurs when power leaks from one channel to the other in systems where channels are not fully isolated.
ANSI/ASA S3.6-2010 regulates the maximum allowable levels of crosstalk for a compliant audiometer. According to this standard, crosstalk must either be less than OdB HL (which means that is not detectable by an average person with normal hearing) or at least 70dB SPL quieter than the test signal transmitting to the other ear.
While purpose-built audiometers meet this standard through the use of hardware designed to electrically isolate the left and right channels, a software audiometer running on a computer or mobile device needs to take extra steps in order to meet the standard. A common approach is to augment the computer or mobile device with external hardware, usually called a DAC, that is custom-designed to generate sound in accordance with the ANSI
standard.
This application describes methods on implementing a crosstalk-cancellation system entirely in software allowing the use of audio hardware that does not fully isolate the left and right channels. It is also possible to build equivalent external hardware that is designed to remove crosstalk from an audio system using the same basic novel technique as described in this application.
Definitions Audiometer - A machine used for evaluating hearing loss.
Decibel (dB) - Used in audiometry when referring to sound levels. The decibel is a logarithmic unit that expresses the ratio of two values of a physical quantity. It is important to understand, for example, that a change in volume from 20dB to 30dB represents 10 times the power level, and a change from 20dB
to 40dB represents 100 times the power level.
COMPUTER-BASED AUDIOMETER
A crosstalk reduction system and method is disclosed for use in a computer-based audiometer. Crosstalk which couples between channels is reduced through use of a software module which modifies digital samples being sent to a digital-to-analog converter.
Crosstalk, the unwanted transmission of signals between communication channels, in the context of hearing testing means that an unwanted audible signal is present on one earphone when the test signal is being directed to the earphone of the opposite side.
Amplifier Jack/Plug Headphones Left IL '4 M1 g r _________________________________________________________________ RI
Rig < <
Right Ground Audible crosstalk occurs when power leaks from one channel to the other in systems where channels are not fully isolated.
ANSI/ASA S3.6-2010 regulates the maximum allowable levels of crosstalk for a compliant audiometer. According to this standard, crosstalk must either be less than OdB HL (which means that is not detectable by an average person with normal hearing) or at least 70dB SPL quieter than the test signal transmitting to the other ear.
While purpose-built audiometers meet this standard through the use of hardware designed to electrically isolate the left and right channels, a software audiometer running on a computer or mobile device needs to take extra steps in order to meet the standard. A common approach is to augment the computer or mobile device with external hardware, usually called a DAC, that is custom-designed to generate sound in accordance with the ANSI
standard.
This application describes methods on implementing a crosstalk-cancellation system entirely in software allowing the use of audio hardware that does not fully isolate the left and right channels. It is also possible to build equivalent external hardware that is designed to remove crosstalk from an audio system using the same basic novel technique as described in this application.
Definitions Audiometer - A machine used for evaluating hearing loss.
Decibel (dB) - Used in audiometry when referring to sound levels. The decibel is a logarithmic unit that expresses the ratio of two values of a physical quantity. It is important to understand, for example, that a change in volume from 20dB to 30dB represents 10 times the power level, and a change from 20dB
to 40dB represents 100 times the power level.
2 dB SPL - Sound Pressure Level: The local pressure deviation from the ambient (average, or equilibrium) atmospheric pressure, caused by a sound wave. A sound meter displays the "loudness"
of a sound with this unit.
RETSPL - Reference Equivalent Threshold Sound Pressure Level. This is the minimum sound level (measured in dB SPL) that a normal person can detect. Audiometers need RETSPL values for each frequency being tested. RETSPL values are specific to each type of headphone and are published for common headphones.
dB HL - Hearing Level. This unit is used to label the "volume control" knob on audiometers, and is also used on the Y-axis of an audiogram to denote the hearing thresholds of the patient. 0 dB HL represents the point at which the human ear can no longer hear the sound. (If you are using TDH style headphones at 2000 Hz, the 0 dB HL level is equal to 11 dB SPL since the published (in the ANSI standard) RETSPL value at 2000 Hz is 11).
ANSI/ASA S3.6-2010 - The most recent specification for audiometers (see: http: //webstore.ansi.org / RecordDetail.aspx?
sku=ANSI%2FASA+S3.6-2010) Relationship between RETSPL and dB HL - Unlike sound meters, the human ear is not equally sensitive to all sound frequencies.
Thus 0 dB SPL (as read on a sound meter) does not represent "no sound" for a human ear. 0 dB HL is higher than 0 dB SPL but by an amount that varies by frequency. The number of decibels to add to 0 dB SPL to reach 0 dB HL is specific to the headphone used and the frequency. This is the RETSPL value for the frequency.
of a sound with this unit.
RETSPL - Reference Equivalent Threshold Sound Pressure Level. This is the minimum sound level (measured in dB SPL) that a normal person can detect. Audiometers need RETSPL values for each frequency being tested. RETSPL values are specific to each type of headphone and are published for common headphones.
dB HL - Hearing Level. This unit is used to label the "volume control" knob on audiometers, and is also used on the Y-axis of an audiogram to denote the hearing thresholds of the patient. 0 dB HL represents the point at which the human ear can no longer hear the sound. (If you are using TDH style headphones at 2000 Hz, the 0 dB HL level is equal to 11 dB SPL since the published (in the ANSI standard) RETSPL value at 2000 Hz is 11).
ANSI/ASA S3.6-2010 - The most recent specification for audiometers (see: http: //webstore.ansi.org / RecordDetail.aspx?
sku=ANSI%2FASA+S3.6-2010) Relationship between RETSPL and dB HL - Unlike sound meters, the human ear is not equally sensitive to all sound frequencies.
Thus 0 dB SPL (as read on a sound meter) does not represent "no sound" for a human ear. 0 dB HL is higher than 0 dB SPL but by an amount that varies by frequency. The number of decibels to add to 0 dB SPL to reach 0 dB HL is specific to the headphone used and the frequency. This is the RETSPL value for the frequency.
3 Sine Wave - The sine wave or sinusoid is a mathematical curve that describes a smooth repetitive oscillation and is defined by: a * sin (wt 4- phase ) = a * sin(2nft + phase) where:
a = amplitude co - angular frequency - 2n * f, where f is frequency t = time phase = the phase, specifies (in radians) where in its cycle the oscillation is at t = 0.
Test signal channel - The audio output channel going to the ear being tested.
Opposite channel - The audio output channel going to the ear that is not being tested.
Amplitude - the objective measurement of the degree of change (positive or negative) in atmospheric pressure (the compression and rarefaction of air molecules) caused by sound waves.
Sample - A value at a specific point in time. In the field of audiology this is referring to a calculated value of the sine wave at a given point in time.
Sample Rate - Specifies the frequency of samples to produce a digital audio signal.
Components Test Signal Sample Generator - Produces a stream of digital samples which are sent to a digital-to-analog converter. It is configured with various parameters which affect the
a = amplitude co - angular frequency - 2n * f, where f is frequency t = time phase = the phase, specifies (in radians) where in its cycle the oscillation is at t = 0.
Test signal channel - The audio output channel going to the ear being tested.
Opposite channel - The audio output channel going to the ear that is not being tested.
Amplitude - the objective measurement of the degree of change (positive or negative) in atmospheric pressure (the compression and rarefaction of air molecules) caused by sound waves.
Sample - A value at a specific point in time. In the field of audiology this is referring to a calculated value of the sine wave at a given point in time.
Sample Rate - Specifies the frequency of samples to produce a digital audio signal.
Components Test Signal Sample Generator - Produces a stream of digital samples which are sent to a digital-to-analog converter. It is configured with various parameters which affect the
4 characteristics of the output e.g. frequency, amplitude, and phase.
Calibration Parameters Lookup Module - A module, that for specified frequency and hearing level values, returns parameters with which the Test Signal Sample Generator should be configured in order to produce an output with the specified frequency and hearing level.
Two Channel Digital-to-Analog Converter - A hardware component of the computer-based audiometer which consumes digital data and produces audio output.
Procedure to generate a tone First the reader will need to become familiar with the procedure to generate a simple sine wave signal at a specified frequency and phase. The basic sine wave (without worrying about phase shift for now) is produced by using the following Sine Wave formula: a * sin(2nft). The frequency (f) and amplitude (a) remain fixed. When digitally generating a sine wave, the time (t) is quantized into intervals called samples. Only the time (t) variable changes for each sample point along the length of the wave. In the above formula, t is replaced by an interval-dependent time by defining time = frequency / sample_rate *
interval.
So, for example, if the frequency is 1Hz and the sample rate is 30 per second, then 30 samples are needed for 1 second of sound. When finding the 5th (index 4) of the 30 samples (numbered 0 to 29), we would thus set t = 1 / 30 * 4.
Calibration Parameters Lookup Module - A module, that for specified frequency and hearing level values, returns parameters with which the Test Signal Sample Generator should be configured in order to produce an output with the specified frequency and hearing level.
Two Channel Digital-to-Analog Converter - A hardware component of the computer-based audiometer which consumes digital data and produces audio output.
Procedure to generate a tone First the reader will need to become familiar with the procedure to generate a simple sine wave signal at a specified frequency and phase. The basic sine wave (without worrying about phase shift for now) is produced by using the following Sine Wave formula: a * sin(2nft). The frequency (f) and amplitude (a) remain fixed. When digitally generating a sine wave, the time (t) is quantized into intervals called samples. Only the time (t) variable changes for each sample point along the length of the wave. In the above formula, t is replaced by an interval-dependent time by defining time = frequency / sample_rate *
interval.
So, for example, if the frequency is 1Hz and the sample rate is 30 per second, then 30 samples are needed for 1 second of sound. When finding the 5th (index 4) of the 30 samples (numbered 0 to 29), we would thus set t = 1 / 30 * 4.
5 The system is designed to use an "audio output framework"
typically provided by either the computer's operating system (OS) or a sound generation library. The audio output framework requires a set of samples in an array-like structure and it takes care of producing an audible analogue audio signal from the given digital sound samples. There may be variations between audio output frameworks for the method of requesting and filling the audio sample buffers. For example some poll the application when the sample buffer requires more data, and some rely on the application to actively send new samples on a regular basis. The same system for creating the buffer applies in either case however.
The pseudocode below shows how to create an array of samples to pass to the operating system to produce a sine wave. Variables that control the maximum amplitude and frequency of the waveform are controlled outside of the loop. The loop itself iterates, at a calculated interval (ie. sample rate), over one full sine wave cycle.
for interval = 0; interval < sampleRate; interval++
var time = frequency / sampleRate * interval sample[interval] = amplitude * sin ( 2 * pi * frequency *
time ) If we graph using this algorithm (in this case setting the frequency to 1Hz, the amplitude to 0.4, and the sample rate to per second) the following wave is generated:
typically provided by either the computer's operating system (OS) or a sound generation library. The audio output framework requires a set of samples in an array-like structure and it takes care of producing an audible analogue audio signal from the given digital sound samples. There may be variations between audio output frameworks for the method of requesting and filling the audio sample buffers. For example some poll the application when the sample buffer requires more data, and some rely on the application to actively send new samples on a regular basis. The same system for creating the buffer applies in either case however.
The pseudocode below shows how to create an array of samples to pass to the operating system to produce a sine wave. Variables that control the maximum amplitude and frequency of the waveform are controlled outside of the loop. The loop itself iterates, at a calculated interval (ie. sample rate), over one full sine wave cycle.
for interval = 0; interval < sampleRate; interval++
var time = frequency / sampleRate * interval sample[interval] = amplitude * sin ( 2 * pi * frequency *
time ) If we graph using this algorithm (in this case setting the frequency to 1Hz, the amplitude to 0.4, and the sample rate to per second) the following wave is generated:
6 oso Inputs: ampu0.4, fresrl Hz Formula:
Constants: sample_rate-30 level = amp * IN (2 * pi * fret' sample_rate * index) 0.162695 = 0.4*SIN(2*pi*1/30*2) 411µ
0.10 ID
X
To e 3 4 6 7 6 3 SAM11221 'mac, 17a.,9 -0.297258 = 0.4*SIN(2*pi*1/30*19) Ti me For most applications you would not use a sample rate as low as 30Hz of course. A digital audio sample rate of 44.1kHz is more 5 typical and so requires 44100 samples to be calculated for generating 1 second of sound.
Phase Shift 10 The basic formula is a * sin(2nft + phase), but if we only need to support a phase shift of 180 degrees, the formula can be simplified to just a * sin(21ift) * -1 as illustrated by the following modified pseudocode:
Constants: sample_rate-30 level = amp * IN (2 * pi * fret' sample_rate * index) 0.162695 = 0.4*SIN(2*pi*1/30*2) 411µ
0.10 ID
X
To e 3 4 6 7 6 3 SAM11221 'mac, 17a.,9 -0.297258 = 0.4*SIN(2*pi*1/30*19) Ti me For most applications you would not use a sample rate as low as 30Hz of course. A digital audio sample rate of 44.1kHz is more 5 typical and so requires 44100 samples to be calculated for generating 1 second of sound.
Phase Shift 10 The basic formula is a * sin(2nft + phase), but if we only need to support a phase shift of 180 degrees, the formula can be simplified to just a * sin(21ift) * -1 as illustrated by the following modified pseudocode:
7 var inversion = -1 // Set to 0 for phase of 0 degrees, or -1 for phase of 180 degrees.
for interval = 0; interval < sampleRate; interval++
var time = frequency / sampleRate * interval sample[interval] = amplitude * sin ( 2 * pi * frequency *
time ) * inversion Procedure to calibrate volume levels As a prerequisite to describing the process for eliminating the cross-talk signal from the Opposite Channel, we must first describe how the system achieves an accurate volume level for each frequency on the Test Signal Channel.
Every headphone, even of the same model, can produce a different sound output level, even when driven at exactly the same voltage level.
For this reason, we must calibrate the power output levels specifically for each headphone at each used frequency while connected to a sound meter to ensure that the correct volume is produced.
In any programmable computer device capable of generating sound, there are usually two factors which combine to produce the resulting sound. The first is the Signal Level. This is the amplitude of the waveform generated or played by the software.
The second is the Power Level. This is the hardware "volume control" of the device which physically amplifies or dampens the signal.
The most straight-forward way is to build a table that maps the desired output level (db SPL) to the required levels of each of
for interval = 0; interval < sampleRate; interval++
var time = frequency / sampleRate * interval sample[interval] = amplitude * sin ( 2 * pi * frequency *
time ) * inversion Procedure to calibrate volume levels As a prerequisite to describing the process for eliminating the cross-talk signal from the Opposite Channel, we must first describe how the system achieves an accurate volume level for each frequency on the Test Signal Channel.
Every headphone, even of the same model, can produce a different sound output level, even when driven at exactly the same voltage level.
For this reason, we must calibrate the power output levels specifically for each headphone at each used frequency while connected to a sound meter to ensure that the correct volume is produced.
In any programmable computer device capable of generating sound, there are usually two factors which combine to produce the resulting sound. The first is the Signal Level. This is the amplitude of the waveform generated or played by the software.
The second is the Power Level. This is the hardware "volume control" of the device which physically amplifies or dampens the signal.
The most straight-forward way is to build a table that maps the desired output level (db SPL) to the required levels of each of
8 the 2 input factors.
It is straightforward to see that for any desired sound output level, there are many possible combinations of the two inputs that can produce the desired output level.
The first idea that might dcome to mind is to always leave the Power Level at 50% and just vary your signal level from say 0%
to 100%.
The problem with this is that you'll likely find you can't achieve the desired sound range.
At 50% power level, with the signal level at 0%, the ground noise (ie. "hiss") from the amplifier will limit how quiet the output can get, and with the signal level at 100%, the maximum output level will likely not be loud enough to meet requirements. The same type of problems occur with the power level permanently set to 0% or 100%.
The next idea you might try is to choose a handful of Power Levels (say 0%, 20%, 40%, ..., %100) with various signal levels for each, so that all desired output levels can be reached. This certainly can work, but there's a consequence. The table that you build would have Signal Levels increasing for each desired output level until you reach the next Power Level increase, at which point the signal level will start back at a lower value than the previous one. This means that the signal value increases non-linearly across the volume range. For example, if the 35dB and 36dB straddled a Power Level step, the signal level could be higher for 35dB than for 36dB. Although this can work perfectly well to produce the right output level, it is not recommended since it makes crosstalk cancellation more complex because power level adjustment (a physical characteristic of the hardware) is inherently nonlinear and doesn't follow a calculable trajectory.
It is straightforward to see that for any desired sound output level, there are many possible combinations of the two inputs that can produce the desired output level.
The first idea that might dcome to mind is to always leave the Power Level at 50% and just vary your signal level from say 0%
to 100%.
The problem with this is that you'll likely find you can't achieve the desired sound range.
At 50% power level, with the signal level at 0%, the ground noise (ie. "hiss") from the amplifier will limit how quiet the output can get, and with the signal level at 100%, the maximum output level will likely not be loud enough to meet requirements. The same type of problems occur with the power level permanently set to 0% or 100%.
The next idea you might try is to choose a handful of Power Levels (say 0%, 20%, 40%, ..., %100) with various signal levels for each, so that all desired output levels can be reached. This certainly can work, but there's a consequence. The table that you build would have Signal Levels increasing for each desired output level until you reach the next Power Level increase, at which point the signal level will start back at a lower value than the previous one. This means that the signal value increases non-linearly across the volume range. For example, if the 35dB and 36dB straddled a Power Level step, the signal level could be higher for 35dB than for 36dB. Although this can work perfectly well to produce the right output level, it is not recommended since it makes crosstalk cancellation more complex because power level adjustment (a physical characteristic of the hardware) is inherently nonlinear and doesn't follow a calculable trajectory.
9 Instead it's best to start with the Signal Level at maximum for the whole volume range, and then choose the Power Level which gets you as close to, but not less than, the target output level. Once at the right power level, the signal level is then adjusted downward (aka "attenuated") by the smallest amount possible until the desired sound level is reached. While it is still true that the signal level still goes up and down slightly throughout the output range, we found that this resulted in the most even and predictable steps, and made cross-talk compensation easier to handle.
Process of building a table for the device power levels When using typical consumer hardware, the device power level attenuations (ie. "the effect of the physical volume control buttons") are not known ahead of time. In order to have a method of producing accurate volume levels there needs to be a way for the application to know the device output power level attenuation for a given system volume level value. This may be recorded by using a sound meter if headphones or speakers are attached. However, since doing the test acoustically would introduce headphones (with potentially unknown properties), it is critical to determine the attenuation effect of the device's volume control hardware by directly measuring the output of the audio jack electrically on equipment capable of accurately measuring microvolt changes.
When using systems that present the volume control API as floating point number between 0.0 and 1.0, it may be necessary to first quantize this range into a countable set. This can be done by experimentally increment through the range from 0.0 to 1.0 at some resolution (say by 0.00001 steps) and look for changes in the output to determine how many significant steps are available.
In the example below, the values were measured using a software oscilloscope running on a laptop. The measurement was conducted by changing the system volume level (controlled in this case by a floating point number between 0 and 1) in small increments starting from maximum (1.000000), and working downwards to minimum (0.000000) while maintaining the signal amplitude at 100%. The result of the measurement is a table where system volume control levels are mapped to an output "power level attenuation" (measured in dB since, as per the definition of dB, we want to measure the relative change in voltage from maximum).
Table: Example of a "System Volume to Power Level Attenuation Mapping" table Index System Volume Power Level Attenuation (dB) 0 1.000000 0 1 0.986000 -0.48 2 0.977000 -1.01 3 0.967000 -1.49 4 0.958000 -2.01 5 0.948000 -2.5 0.000000 -115.45 By using this "System Volume to Power Level Attenuation Mapping"
table, the application now has the information needed to produce a signal that is for example 2.01 dB below the device maximum output level by setting the system volume to 0.958000 along with a signal amplitude of 1 (ie. 100% when the amplitude range is from 0 to 1).
NOTE: Building a table of "relative change" (in dB) starting from 0 and working upwards would have been impossible. We always base everything from the maximum possible output, which is why this table starts from the maximum output and works downwards.
NOTE: This table is still referring to how the volume control buttons affect the output voltage level only. The next step is of course to attach headphones to see how loud of a sound can be produced (in dB SPL) at maximum power output.
The maximum dB
SPL is of course is headphone dependent, but with good headphones that respond linearly to voltage change, the sound level will attenuate according to the values in the Power Level Attenuation column.
The above table allows the application to have the information to produce precise sound levels, but only if they fall exactly on the intervals (ie rows) in the table. To get more resolution and range, we also need to combine this with attenuation of the signal itself.
The graph below illustrates the how, in isolation, neither the power level attenuation nor signal level attenuation would give enough range but when combining them together will allow the application to achieve a high signal to noise ratio over a large range.
NOTE: The green line has the most range; It can produce very quiet sounds (bottom-left), while still being able to produce very loud sounds (top-right).
Chart: Device Power Level Attenuation (blue), Signal Level Attenuation (red), and "Power Level Attenuation + Signal Level Attenuation" (green) 0 --device volume - attenuation dB
-50 ¨ signal attenuation dB
device volume -100 attenuation dB + signal attenuation dB
0 02 0.4 OA 0.8 1 Process of adjusting the signal level When a target output attenuation (dB) value is requested by the user, the application needs to determine the required system volume level and also the required signal attenuation to achieve the requested output level. This is done by using the lookup table to find the closest power level attenuation value that attenuates the output as close as possible to (but not less than) the target output attenuation. Once the power level attenuation has been read from the table, the application then needs to calculate how much additional signal attenuation is required to achieve the target attenuation. This formula is simple:
Signal attenuation dB = Target Attenuation dB - Power Level Attenuation dB
Example:
Let's say the target attenuation is -1.7 dB. The application would use the lookup table to find the lowest power level attenuation value which is bigger than the target attenuation.
In our example this would be -1.49dB (index 3 at the table above). By using the formula above the signal attenuation would be following:
Signal attenuation dB = -1.7 dB - -1.49dB = -0.21dB
Thus the remaining attenuation that is needed on the signal itself is -0.21dB. This needs to be converted from decibels into amplitude in order to use this value to adjust the sine wave (which needs an amplitude value between 0 and 1).
A helper methods let us do that conversion easily:
func decibelstoAmplitude(dB) {
return pow(10, dB/20) Combining all of the above concepts, we can finally show the pseudocode to lookup the required system power level and calculate the required signal attenuation for any requested target attenuation.
func calculateSignalAndPower(targetAttenuation) f for i = 1; i < systemVolumePowerLevelTable.count; i++
powerLevelAttenuation = systemVolumePowerLevelTable[i].
powerLevelAttenuation if (targetAttenuation > powerLevelAttenuation) /* The target was greater than the powerLevelAttenuation, use the previous one and calculate the delta */
var systemVolumeLevel = systemVolumePowerLevelTable[i-1] .
systemVolumeLevel powerLevelAttenuation =
systemVolumePowerLevelTable[i-1].powerLevelAttenuation var amplitude = decibelstoAmplitude(targetAttenuation -powerLevelAttenuation) return f systemVolumeLevel, amplitude 1 The next step is to use the 2 values returned by this function (systemVolumeLevel and amplitude) to generate the sound:
// Set the system volume level (The actual API will be OS-dependent of course) system.setVolumeLevel( systemVolumeLevel ) // Generate the waveform samples at the desired amplitude for interval = 0; interval < sampleRate; interval++
var time = frequency / sampleRate * interval sample[interval] = amplitude * sin ( 2 * pi * frequency *
time ) The "sample" array now contains the data required to pass to the sound generation module.
Finding the Attenuation for HL
In a hearing test application, it is necessary to be able to ask for sounds to be produced at a specific level measured in dB
HL. The dB HL to dB SPL conversion is simple and uses a RETSPL
value specific to the frequency as described earlier.
Even so, the sound level produced in dB SPL still depends on the specific transducer that is used. Recall that the device power levels were measured electrically without a transducer attached (in order build a generic system that allows any type of headphones to be calibrated for use with it).
Thus the application cannot directly use the SPL values in calculations. The most obvious way to convert our table of attenuations (in dB) into actual sound levels (in dB SPL) is to use a sound meter to measure the maximum SPL with a transducer connected and by playing a tone at maximum power level with maximum amplitude at each frequency.
Once the "real world" maximum sound levels are known, the target attenuation can be calculated by using the following formula:
targetSPL = HL + RETSPL
targetAttenuation = targetSPL - maxSPL
Due to the variable nature of the transducers, the approach of calculating the target attenuation based on one globally defined maximum SPL is not valid.
Instead, each transducer must be calibrated. In real terms, this just means noting the peculiar maximum level of each transducer as measured on a sound meter. Other variants of this can be done which use a globally defined maximum but then introduce an "adjustment" value which is customized for each headphone to shift the mapping up or down slightly as needed. This process is referred to as "calibrating a transducer".
Process for finding the correct adjustment value would be the following:
var adjustment = 0 do targetAttenuation = targetSPL + adjustment - maxSPL
calculateSignalAndPower(targetAttenuation) system.setVolumeLevel( systemVolumeLevel ) for interval = 0; interval < sampleRate; interval++
var time = frequency / sampleRate * interval sample[interval] = amplitude * sin ( 2 * pi *
frequency * time ) // measure SPL value from the sound meter adjustment = targetSPL - measuredSPL
while targetSPL != measuredSPL
Crosstalk Cancellation Method The following describes a system of compensating for predictable cross talk. Using a sound meter, the crosstalk in the system is measured (for each frequency at each volume level), and that data is then recorded into the application. Later when in operation, these numbers are used to generate a signal which counteracts the crosstalk. The process is able to compensate for crosstalk that is at the same frequency as the signal but at a different volume level and at a phase that is either approximately in phase or 180 degrees out of phase.
Signal on test channel 1.30 O.
\\\
0.3 To. C Z 3 4 5 E.1 1! '2 3 '= IS 19 ZO 22 n 24 2.5 26 27 n 21 -am /14r \\\
Crosstalk on opposite channel, in this case being 180 degrees Ø40 out of phase with test signal.
Time For a given frequency, the crosstalk has been found to flip to the opposite phase at higher volumes as compared to the phase manifested at lower volumes. The level of crosstalk has also been found to vary in a non-linear fashion as volume increases. The solution described here handles compensation of both non-linear and phase-inconsistent crosstalk in a system.
In this method, cancelling the crosstalk signal is based on measuring the output of the test channel and then measuring the output of the opposite channel (where the crosstalk is observed). The required crosstalk compensation signal can then be calculated as follows:
crosstalkSignalAttenuation = oppositeChannelSPL -testSignalChannelSPL
The crosstalkSignalAttenuation can now be used to calculate the required signal amplitude. Of course, we don't want to adjust the power level of the system at this point because this would have the side effect of altering the test signal. Instead we just have to work with whatever power level is currently set to. All the attenuation must be done on the signal only.
The test signal and crosstalk cancellation signal should be produced by the same code module which sends two channel samples to audio hardware, so that there is high degree of accuracy to the time synchronization between the two channels.
As an example, if the test signal is measured at 75dB
(measuredSPL) and the crosstalk is measured at 25dB
(measuredCrosstalkSPL), we would do the following:
// First calculate the system volume and amplitude for the test signal struct {systemVolumeLevel, amplitude} = calculateSignalAndPower (targetAttenuation) // Set the system volume level according to the test signal system.setVolumeLevel( systemVolumeLevel ) // Generate the waveform samples at the desired amplitude for the test signal for interval = 0; interval < sampleRate; interval++
var time - frequency / sampleRate * interval sample[interval] = amplitude * sin ( 2 * pi * frequency *
time ) // Calculate amplitude for the crosstalk signal.
// Note: We can only attenuate signal, but not adjust system volume to achieve crosstalk compensation crosstalkSignalAttenuation = measuredCrosstalkSPL + adjustment -measuredSPL
crosstalkAmplitude = decibelstoAmplitude (crosstalkSignalAttenuation) // Generate the waveform samples at the desired amplitude for the test signal for interval = 0; interval < sampleRate; interval++
var time frequency / sampleRate * interval crossTalkSample[interval] = crosstalkAmplitude * sin (2*pi*frequency*time) [
Process of Detecting the Crosstalk Signal Phase As described earlier, the approach described here assumes the crosstalk signal either being in phase (0 degree shift) or opposite phase (180 degree shift). After the crosstalk cancellation signal has been produced, the required phase shift can be inferred by measuring the cancelled crosstalk SPL and then producing the same signal but with the opposite phase. If the measured crosstalk SPL level goes down, then the crosstalk signal has the opposite phase from the test signal.
// Measure the opposite channel SPL after the adjustments above crosstalkSignalAttenuationInPhase = measuredCrosstalkSPL - measuredSPL
var phaseInversion = -1 // Generate the waveform samples at the desired amplitude for the test signal for interval = 0; interval < sampleRate; interval++
var time = frequency / sampleRate * interval crossTalkSample[interval] = crosstalkAmplitude *
sin(2*pi*frequency*time) * phaseInversion // Measure the opposite channel SPL again after adjusting the phase crosstalkSignalAttenuationOppositePhase = measuredCrosstalkSPL -measuredSPL
if crosstalkSignalAttenuationOppositePhase crosstalkSignalAttenuationInPhase) // The crosstalk signal is on opposite phase to the test signal for the measured HL
else // The crosstalk signal is in same phase as the test signal for the measured HL
Crosstalk Signal Calibration Flowchart The following flowchart describes the steps for finding the correct crosstalk cancellation signal level. The steps may be performed by a person manually adjusting the values or they may be conducted by an automated system capable of measuring sound levels and adjusting the values.
t Calibration /tart Crosstall) Measure Test Procedure Cancellation I. Play Tone Ir-Channel Calibratbn Output SPL
' V
Measure r 0 Crosstalk SPL
."---- ---...\
on the Other IP
r Channel --- ---Hearing Level Calibration Database ..,...
= .
, ., \---. _____________ ---j ss ss, , , , Calculate New Adjustment Based on i Crosstalk Measurement And Test Channel Output - re there pd..
s Crosstalk i calculated NO
Within Allowed ,djustments Limits?
, I
I YES
' NO
I
i i 1. Reached YES
Maximum djustments YES
..
Switch Is Phase Shift NO s Crosstalk Tested?
Signal Phase YES (we have found the lowest value) i ....
,i, Crosstalk Cancellation -, Calibration Completed [
Storing the crosstalk compensation values in a data structure The "shape" of the crosstalk encountered varies by a combination of factors:
= The headphones being used (aka "Transducer") = The channel (left or right) being tested = The frequency of the test signal = The volume of the test signal (aka "Hearing Level") For these reasons, we must store each experimentally discovered Crosstalk Attenuation value and Phase along with a record of the transducer that was used, the frequency, the volume level, and the channel. Below is an example of a how a database table might be look when storing this data:
Crosstalk Signal Adjustment Table Transducer Frequency Hearing Channel Crosstalk Phase Level Attenuation TDH-50 250 50 Left -46.1 1 TDH-50 250 50 Right -48.0 1 ===
TDH-50 500 95 Left -23.6 -1 TDH-50 500 95 Right -22.7 -1 Crosstalk Signal Generation Flowchart The following flowchart illustrates the steps required for producing the crosstalk cancellation signal. The chart can be divided into three sections: getting inputs, finding the required system and signal levels and finally producing the samples.
This makes use of the Crosstalk Signal Adjustment Table described above:
i Generation ' Test Channel.
May Tore pn,..d _______________________________________________ b FrequonCY=
kleering LeveL .
Tmnsducer - _ . , =
= =
CalcubX Test Chennel Target SPL usng ¨ ____ --......, Fond System (---= RETSPL and -I_ 4__________ 1s-.
4 Power Level 4-4-4' Herein LevelFind Sarni k:...: - - ¨ ---.:, e..... ====.µ ....- tor Tweet Actusesers Level -=- - =-----AAustment tor ...
totering Level Z....= --------"::, -....-z....._,....---,-...Haulm Levet '' = =
.- ". . =.' Database Calculate Signal System Power to - .- r -".. ..=
------ Atteuston based ', AftanuaTaan - --- ...= " -------------- a on Target SPL .
Database - ' -. .' and System Fire Crosstalk /
Power Level =====
AdAstment tor =
.,......
Heaeog Level = - &anal Attervallen 4-- . - ---___ _____.=
' Calculate ________________ based On Crosstalk Signal Signal A4ustment = s=¨to Arnoilluie end %Worn Ob.*, Based on ND Laval Attenuation Other Channel 4 __ I
k __________________________________________________________ Calculate Crosstak Signal Test Chtnnet Arnpltude Bassi on Me Anentiason ¨,-----I
=
=
= __________________ = ________________________________________ =
=
=
Set Systu- f .
. Power Lord Set Swot I __ I Set Sigial = Amplitude and N I
mow. and TFrequency For= ........ otfreheque, cncyhanFr:
...
est Channel =
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.
' ' ' = = . Output Tone Data Structure 1 - - ' = == ...
'...
"-...:\ Test Channet Frequency, 1 = Start Audo Praire Work 4 = -:µ
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(Play Tore) Stopped 1 =
r Summary The crosstalk compensation system described above is capable of handling nonlinear or even completely unpredictable crosstalk with a varying signal phase. No other audiometer has a software cross-talk compensation system. Purpose-built audiometer hardware doesn't need anti-crosstalk algorithms however because the hardware is designed in such a way as to avoid crosstalk. For audiometers built on a generic mobile platform such as an iOS or Android device however, the hardware has typically not been designed to minimize crosstalk to within ANSI
standard, which leads to the need for our software-based crosstalk cancellation technique.
This is not the same as active-noise-cancellation which relies on a feedback loop driven by directly sampling the sound to be cancelled.
In our situation we do not have direct access to sampling the undesired sound so we must predictively cancel the sound rather than reactively cancel it. Cross-talk shape varies depending on the hardware used (ie. mobile device, cables and audio connectors, transducers), and varies with changes to either the source signal volume or frequency. There is no single shape for the crosstalk signal that will globally cancel crosstalk. The solution relies on calibrating values for each individual configuration.
Process of building a table for the device power levels When using typical consumer hardware, the device power level attenuations (ie. "the effect of the physical volume control buttons") are not known ahead of time. In order to have a method of producing accurate volume levels there needs to be a way for the application to know the device output power level attenuation for a given system volume level value. This may be recorded by using a sound meter if headphones or speakers are attached. However, since doing the test acoustically would introduce headphones (with potentially unknown properties), it is critical to determine the attenuation effect of the device's volume control hardware by directly measuring the output of the audio jack electrically on equipment capable of accurately measuring microvolt changes.
When using systems that present the volume control API as floating point number between 0.0 and 1.0, it may be necessary to first quantize this range into a countable set. This can be done by experimentally increment through the range from 0.0 to 1.0 at some resolution (say by 0.00001 steps) and look for changes in the output to determine how many significant steps are available.
In the example below, the values were measured using a software oscilloscope running on a laptop. The measurement was conducted by changing the system volume level (controlled in this case by a floating point number between 0 and 1) in small increments starting from maximum (1.000000), and working downwards to minimum (0.000000) while maintaining the signal amplitude at 100%. The result of the measurement is a table where system volume control levels are mapped to an output "power level attenuation" (measured in dB since, as per the definition of dB, we want to measure the relative change in voltage from maximum).
Table: Example of a "System Volume to Power Level Attenuation Mapping" table Index System Volume Power Level Attenuation (dB) 0 1.000000 0 1 0.986000 -0.48 2 0.977000 -1.01 3 0.967000 -1.49 4 0.958000 -2.01 5 0.948000 -2.5 0.000000 -115.45 By using this "System Volume to Power Level Attenuation Mapping"
table, the application now has the information needed to produce a signal that is for example 2.01 dB below the device maximum output level by setting the system volume to 0.958000 along with a signal amplitude of 1 (ie. 100% when the amplitude range is from 0 to 1).
NOTE: Building a table of "relative change" (in dB) starting from 0 and working upwards would have been impossible. We always base everything from the maximum possible output, which is why this table starts from the maximum output and works downwards.
NOTE: This table is still referring to how the volume control buttons affect the output voltage level only. The next step is of course to attach headphones to see how loud of a sound can be produced (in dB SPL) at maximum power output.
The maximum dB
SPL is of course is headphone dependent, but with good headphones that respond linearly to voltage change, the sound level will attenuate according to the values in the Power Level Attenuation column.
The above table allows the application to have the information to produce precise sound levels, but only if they fall exactly on the intervals (ie rows) in the table. To get more resolution and range, we also need to combine this with attenuation of the signal itself.
The graph below illustrates the how, in isolation, neither the power level attenuation nor signal level attenuation would give enough range but when combining them together will allow the application to achieve a high signal to noise ratio over a large range.
NOTE: The green line has the most range; It can produce very quiet sounds (bottom-left), while still being able to produce very loud sounds (top-right).
Chart: Device Power Level Attenuation (blue), Signal Level Attenuation (red), and "Power Level Attenuation + Signal Level Attenuation" (green) 0 --device volume - attenuation dB
-50 ¨ signal attenuation dB
device volume -100 attenuation dB + signal attenuation dB
0 02 0.4 OA 0.8 1 Process of adjusting the signal level When a target output attenuation (dB) value is requested by the user, the application needs to determine the required system volume level and also the required signal attenuation to achieve the requested output level. This is done by using the lookup table to find the closest power level attenuation value that attenuates the output as close as possible to (but not less than) the target output attenuation. Once the power level attenuation has been read from the table, the application then needs to calculate how much additional signal attenuation is required to achieve the target attenuation. This formula is simple:
Signal attenuation dB = Target Attenuation dB - Power Level Attenuation dB
Example:
Let's say the target attenuation is -1.7 dB. The application would use the lookup table to find the lowest power level attenuation value which is bigger than the target attenuation.
In our example this would be -1.49dB (index 3 at the table above). By using the formula above the signal attenuation would be following:
Signal attenuation dB = -1.7 dB - -1.49dB = -0.21dB
Thus the remaining attenuation that is needed on the signal itself is -0.21dB. This needs to be converted from decibels into amplitude in order to use this value to adjust the sine wave (which needs an amplitude value between 0 and 1).
A helper methods let us do that conversion easily:
func decibelstoAmplitude(dB) {
return pow(10, dB/20) Combining all of the above concepts, we can finally show the pseudocode to lookup the required system power level and calculate the required signal attenuation for any requested target attenuation.
func calculateSignalAndPower(targetAttenuation) f for i = 1; i < systemVolumePowerLevelTable.count; i++
powerLevelAttenuation = systemVolumePowerLevelTable[i].
powerLevelAttenuation if (targetAttenuation > powerLevelAttenuation) /* The target was greater than the powerLevelAttenuation, use the previous one and calculate the delta */
var systemVolumeLevel = systemVolumePowerLevelTable[i-1] .
systemVolumeLevel powerLevelAttenuation =
systemVolumePowerLevelTable[i-1].powerLevelAttenuation var amplitude = decibelstoAmplitude(targetAttenuation -powerLevelAttenuation) return f systemVolumeLevel, amplitude 1 The next step is to use the 2 values returned by this function (systemVolumeLevel and amplitude) to generate the sound:
// Set the system volume level (The actual API will be OS-dependent of course) system.setVolumeLevel( systemVolumeLevel ) // Generate the waveform samples at the desired amplitude for interval = 0; interval < sampleRate; interval++
var time = frequency / sampleRate * interval sample[interval] = amplitude * sin ( 2 * pi * frequency *
time ) The "sample" array now contains the data required to pass to the sound generation module.
Finding the Attenuation for HL
In a hearing test application, it is necessary to be able to ask for sounds to be produced at a specific level measured in dB
HL. The dB HL to dB SPL conversion is simple and uses a RETSPL
value specific to the frequency as described earlier.
Even so, the sound level produced in dB SPL still depends on the specific transducer that is used. Recall that the device power levels were measured electrically without a transducer attached (in order build a generic system that allows any type of headphones to be calibrated for use with it).
Thus the application cannot directly use the SPL values in calculations. The most obvious way to convert our table of attenuations (in dB) into actual sound levels (in dB SPL) is to use a sound meter to measure the maximum SPL with a transducer connected and by playing a tone at maximum power level with maximum amplitude at each frequency.
Once the "real world" maximum sound levels are known, the target attenuation can be calculated by using the following formula:
targetSPL = HL + RETSPL
targetAttenuation = targetSPL - maxSPL
Due to the variable nature of the transducers, the approach of calculating the target attenuation based on one globally defined maximum SPL is not valid.
Instead, each transducer must be calibrated. In real terms, this just means noting the peculiar maximum level of each transducer as measured on a sound meter. Other variants of this can be done which use a globally defined maximum but then introduce an "adjustment" value which is customized for each headphone to shift the mapping up or down slightly as needed. This process is referred to as "calibrating a transducer".
Process for finding the correct adjustment value would be the following:
var adjustment = 0 do targetAttenuation = targetSPL + adjustment - maxSPL
calculateSignalAndPower(targetAttenuation) system.setVolumeLevel( systemVolumeLevel ) for interval = 0; interval < sampleRate; interval++
var time = frequency / sampleRate * interval sample[interval] = amplitude * sin ( 2 * pi *
frequency * time ) // measure SPL value from the sound meter adjustment = targetSPL - measuredSPL
while targetSPL != measuredSPL
Crosstalk Cancellation Method The following describes a system of compensating for predictable cross talk. Using a sound meter, the crosstalk in the system is measured (for each frequency at each volume level), and that data is then recorded into the application. Later when in operation, these numbers are used to generate a signal which counteracts the crosstalk. The process is able to compensate for crosstalk that is at the same frequency as the signal but at a different volume level and at a phase that is either approximately in phase or 180 degrees out of phase.
Signal on test channel 1.30 O.
\\\
0.3 To. C Z 3 4 5 E.1 1! '2 3 '= IS 19 ZO 22 n 24 2.5 26 27 n 21 -am /14r \\\
Crosstalk on opposite channel, in this case being 180 degrees Ø40 out of phase with test signal.
Time For a given frequency, the crosstalk has been found to flip to the opposite phase at higher volumes as compared to the phase manifested at lower volumes. The level of crosstalk has also been found to vary in a non-linear fashion as volume increases. The solution described here handles compensation of both non-linear and phase-inconsistent crosstalk in a system.
In this method, cancelling the crosstalk signal is based on measuring the output of the test channel and then measuring the output of the opposite channel (where the crosstalk is observed). The required crosstalk compensation signal can then be calculated as follows:
crosstalkSignalAttenuation = oppositeChannelSPL -testSignalChannelSPL
The crosstalkSignalAttenuation can now be used to calculate the required signal amplitude. Of course, we don't want to adjust the power level of the system at this point because this would have the side effect of altering the test signal. Instead we just have to work with whatever power level is currently set to. All the attenuation must be done on the signal only.
The test signal and crosstalk cancellation signal should be produced by the same code module which sends two channel samples to audio hardware, so that there is high degree of accuracy to the time synchronization between the two channels.
As an example, if the test signal is measured at 75dB
(measuredSPL) and the crosstalk is measured at 25dB
(measuredCrosstalkSPL), we would do the following:
// First calculate the system volume and amplitude for the test signal struct {systemVolumeLevel, amplitude} = calculateSignalAndPower (targetAttenuation) // Set the system volume level according to the test signal system.setVolumeLevel( systemVolumeLevel ) // Generate the waveform samples at the desired amplitude for the test signal for interval = 0; interval < sampleRate; interval++
var time - frequency / sampleRate * interval sample[interval] = amplitude * sin ( 2 * pi * frequency *
time ) // Calculate amplitude for the crosstalk signal.
// Note: We can only attenuate signal, but not adjust system volume to achieve crosstalk compensation crosstalkSignalAttenuation = measuredCrosstalkSPL + adjustment -measuredSPL
crosstalkAmplitude = decibelstoAmplitude (crosstalkSignalAttenuation) // Generate the waveform samples at the desired amplitude for the test signal for interval = 0; interval < sampleRate; interval++
var time frequency / sampleRate * interval crossTalkSample[interval] = crosstalkAmplitude * sin (2*pi*frequency*time) [
Process of Detecting the Crosstalk Signal Phase As described earlier, the approach described here assumes the crosstalk signal either being in phase (0 degree shift) or opposite phase (180 degree shift). After the crosstalk cancellation signal has been produced, the required phase shift can be inferred by measuring the cancelled crosstalk SPL and then producing the same signal but with the opposite phase. If the measured crosstalk SPL level goes down, then the crosstalk signal has the opposite phase from the test signal.
// Measure the opposite channel SPL after the adjustments above crosstalkSignalAttenuationInPhase = measuredCrosstalkSPL - measuredSPL
var phaseInversion = -1 // Generate the waveform samples at the desired amplitude for the test signal for interval = 0; interval < sampleRate; interval++
var time = frequency / sampleRate * interval crossTalkSample[interval] = crosstalkAmplitude *
sin(2*pi*frequency*time) * phaseInversion // Measure the opposite channel SPL again after adjusting the phase crosstalkSignalAttenuationOppositePhase = measuredCrosstalkSPL -measuredSPL
if crosstalkSignalAttenuationOppositePhase crosstalkSignalAttenuationInPhase) // The crosstalk signal is on opposite phase to the test signal for the measured HL
else // The crosstalk signal is in same phase as the test signal for the measured HL
Crosstalk Signal Calibration Flowchart The following flowchart describes the steps for finding the correct crosstalk cancellation signal level. The steps may be performed by a person manually adjusting the values or they may be conducted by an automated system capable of measuring sound levels and adjusting the values.
t Calibration /tart Crosstall) Measure Test Procedure Cancellation I. Play Tone Ir-Channel Calibratbn Output SPL
' V
Measure r 0 Crosstalk SPL
."---- ---...\
on the Other IP
r Channel --- ---Hearing Level Calibration Database ..,...
= .
, ., \---. _____________ ---j ss ss, , , , Calculate New Adjustment Based on i Crosstalk Measurement And Test Channel Output - re there pd..
s Crosstalk i calculated NO
Within Allowed ,djustments Limits?
, I
I YES
' NO
I
i i 1. Reached YES
Maximum djustments YES
..
Switch Is Phase Shift NO s Crosstalk Tested?
Signal Phase YES (we have found the lowest value) i ....
,i, Crosstalk Cancellation -, Calibration Completed [
Storing the crosstalk compensation values in a data structure The "shape" of the crosstalk encountered varies by a combination of factors:
= The headphones being used (aka "Transducer") = The channel (left or right) being tested = The frequency of the test signal = The volume of the test signal (aka "Hearing Level") For these reasons, we must store each experimentally discovered Crosstalk Attenuation value and Phase along with a record of the transducer that was used, the frequency, the volume level, and the channel. Below is an example of a how a database table might be look when storing this data:
Crosstalk Signal Adjustment Table Transducer Frequency Hearing Channel Crosstalk Phase Level Attenuation TDH-50 250 50 Left -46.1 1 TDH-50 250 50 Right -48.0 1 ===
TDH-50 500 95 Left -23.6 -1 TDH-50 500 95 Right -22.7 -1 Crosstalk Signal Generation Flowchart The following flowchart illustrates the steps required for producing the crosstalk cancellation signal. The chart can be divided into three sections: getting inputs, finding the required system and signal levels and finally producing the samples.
This makes use of the Crosstalk Signal Adjustment Table described above:
i Generation ' Test Channel.
May Tore pn,..d _______________________________________________ b FrequonCY=
kleering LeveL .
Tmnsducer - _ . , =
= =
CalcubX Test Chennel Target SPL usng ¨ ____ --......, Fond System (---= RETSPL and -I_ 4__________ 1s-.
4 Power Level 4-4-4' Herein LevelFind Sarni k:...: - - ¨ ---.:, e..... ====.µ ....- tor Tweet Actusesers Level -=- - =-----AAustment tor ...
totering Level Z....= --------"::, -....-z....._,....---,-...Haulm Levet '' = =
.- ". . =.' Database Calculate Signal System Power to - .- r -".. ..=
------ Atteuston based ', AftanuaTaan - --- ...= " -------------- a on Target SPL .
Database - ' -. .' and System Fire Crosstalk /
Power Level =====
AdAstment tor =
.,......
Heaeog Level = - &anal Attervallen 4-- . - ---___ _____.=
' Calculate ________________ based On Crosstalk Signal Signal A4ustment = s=¨to Arnoilluie end %Worn Ob.*, Based on ND Laval Attenuation Other Channel 4 __ I
k __________________________________________________________ Calculate Crosstak Signal Test Chtnnet Arnpltude Bassi on Me Anentiason ¨,-----I
=
=
= __________________ = ________________________________________ =
=
=
Set Systu- f .
. Power Lord Set Swot I __ I Set Sigial = Amplitude and N I
mow. and TFrequency For= ........ otfreheque, cncyhanFr:
...
est Channel =
JrJr .
.
' ' ' = = . Output Tone Data Structure 1 - - ' = == ...
'...
"-...:\ Test Channet Frequency, 1 = Start Audo Praire Work 4 = -:µ
arrOolude ..,.. Diner Channel: Frequency.
..-arrolonsts .="' --0 --'...
. -. = ' .
II' Fl Samples , = == " - -Moth = ' Channels) =
(Play Tore) Stopped 1 =
r Summary The crosstalk compensation system described above is capable of handling nonlinear or even completely unpredictable crosstalk with a varying signal phase. No other audiometer has a software cross-talk compensation system. Purpose-built audiometer hardware doesn't need anti-crosstalk algorithms however because the hardware is designed in such a way as to avoid crosstalk. For audiometers built on a generic mobile platform such as an iOS or Android device however, the hardware has typically not been designed to minimize crosstalk to within ANSI
standard, which leads to the need for our software-based crosstalk cancellation technique.
This is not the same as active-noise-cancellation which relies on a feedback loop driven by directly sampling the sound to be cancelled.
In our situation we do not have direct access to sampling the undesired sound so we must predictively cancel the sound rather than reactively cancel it. Cross-talk shape varies depending on the hardware used (ie. mobile device, cables and audio connectors, transducers), and varies with changes to either the source signal volume or frequency. There is no single shape for the crosstalk signal that will globally cancel crosstalk. The solution relies on calibrating values for each individual configuration.
Claims (10)
1. A computer-implemented method for reducing an undesired crosstalk signal in an audiometer by software means alone.
2. The computer-implemented method of claim 1 wherein said software means comprises generating a counter signal that is equal in frequency and phase and opposite in amplitude to the undesired crosstalk signal.
3. A computer-implemented method for reducing undesired crosstalk signals on an inactive channel of a device comprising the steps of:
(i) determining system volume levels and associated signal amplitudes required to achieve a range of desired audio output attenuation levels on an active channel of the device;
(ii) determining a crosstalk compensation signal comprising a signal amplitude and associated phase shift required to reduce undesired crosstalk on the inactive channel of the device for each desired audio output attention level in the range of desired audio output attenuation levels; and (iii) generating a desired audio output attenuation level on the active channel of the device by generating a signal at the determined system volume level and associated signal amplitude required to achieve said desired audio output attenuation level, and generating a contemporaneous crosstalk compensation signal on the inactive channel of the device by generating a signal at the determined signal amplitude and associated phase shift required to reduce the undesired crosstalk on the inactive channel.
(i) determining system volume levels and associated signal amplitudes required to achieve a range of desired audio output attenuation levels on an active channel of the device;
(ii) determining a crosstalk compensation signal comprising a signal amplitude and associated phase shift required to reduce undesired crosstalk on the inactive channel of the device for each desired audio output attention level in the range of desired audio output attenuation levels; and (iii) generating a desired audio output attenuation level on the active channel of the device by generating a signal at the determined system volume level and associated signal amplitude required to achieve said desired audio output attenuation level, and generating a contemporaneous crosstalk compensation signal on the inactive channel of the device by generating a signal at the determined signal amplitude and associated phase shift required to reduce the undesired crosstalk on the inactive channel.
4. The computer-implemented method of claim 3 wherein determining system volume levels and associated signal amplitudes required to achieve the range of desired audio output attenuation levels on the active channel of the device comprises the steps of:
(i) measuring an output level at each system volume level of the device from maximum level to minimum level when the associated signal amplitude is set at 1;
(ii) converting the output level at each such system volume level into decibels of attenuation;
(iii) storing the decibels of attenuation for each system volume level in a database;
(iv) querying the database to find the system volume level that attenuates the output level to as close to, but not more than, the desired audio output attenuation level;
(v) calculating an attenuation required on the signal amplitude, which when added to the decibel of attenuation achieved at said system volume level, will result in producing the desired audio output attenuation level;
(vi) storing the calculated attenuation required on the signal amplitude in the database; and (vii) repeating steps (iv) to (vi) for all system volume levels in the database.
(i) measuring an output level at each system volume level of the device from maximum level to minimum level when the associated signal amplitude is set at 1;
(ii) converting the output level at each such system volume level into decibels of attenuation;
(iii) storing the decibels of attenuation for each system volume level in a database;
(iv) querying the database to find the system volume level that attenuates the output level to as close to, but not more than, the desired audio output attenuation level;
(v) calculating an attenuation required on the signal amplitude, which when added to the decibel of attenuation achieved at said system volume level, will result in producing the desired audio output attenuation level;
(vi) storing the calculated attenuation required on the signal amplitude in the database; and (vii) repeating steps (iv) to (vi) for all system volume levels in the database.
5. The computer-implemented method of claim 4 wherein determining the crosstalk compensation signal comprising the signal amplitude and associated phase shift required to reduce undesired crosstalk on the inactive channel of the device for each desired audio output attention level in the range of desired audio output attenuation levels comprises the steps of:
(i) generating a tone for the desired audio output attenuation level by controlling the system volume level and associated signal amplitude required to generate said tone;
(ii) measuring the undesired crosstalk on the inactive channel of the device;
(iii) calculating a signal amplitude that would be required to eliminate the undesired crosstalk on the inactive channel without modifying the system volume level;
(iv) generating a candidate crosstalk compensation signal at a phase of 180 degrees in accordance with step (iii);
(v) re-measuring the undesired crosstalk and iteratively adjusting the associated signal amplitude until the measured undesired crosstalk is minimized;
(vi) generating a candidate crosstalk compensation signal at a phase of 0 degrees in accordance with step (iii);
(vii) re-measuring the undesired crosstalk and iteratively adjusting the associated signal amplitude until the measured crosstalk is minimized; and (viii) comparing the candidate crosstalk compensation signal at a phase of 180 degrees to the candidate crosstalk compensation signal at a phase of 0 degrees to determine which compensation signal is more effective at reducing the undesired crosstalk, and storing the signal amplitude and phase of such more effective crosstalk cancellation signal in a database.
(i) generating a tone for the desired audio output attenuation level by controlling the system volume level and associated signal amplitude required to generate said tone;
(ii) measuring the undesired crosstalk on the inactive channel of the device;
(iii) calculating a signal amplitude that would be required to eliminate the undesired crosstalk on the inactive channel without modifying the system volume level;
(iv) generating a candidate crosstalk compensation signal at a phase of 180 degrees in accordance with step (iii);
(v) re-measuring the undesired crosstalk and iteratively adjusting the associated signal amplitude until the measured undesired crosstalk is minimized;
(vi) generating a candidate crosstalk compensation signal at a phase of 0 degrees in accordance with step (iii);
(vii) re-measuring the undesired crosstalk and iteratively adjusting the associated signal amplitude until the measured crosstalk is minimized; and (viii) comparing the candidate crosstalk compensation signal at a phase of 180 degrees to the candidate crosstalk compensation signal at a phase of 0 degrees to determine which compensation signal is more effective at reducing the undesired crosstalk, and storing the signal amplitude and phase of such more effective crosstalk cancellation signal in a database.
6. The computer-implemented method of claim 5 wherein the step of generating the desired audio output attenuation level on the active channel of the device comprises querying the database for the system volume level and associated signal amplitude required to achieve said desired audio output attenuation level, and the step of generating a contemporaneous crosstalk compensation signal on the inactive channel of the device comprises querying the database for the signal amplitude and phase required to achieve said crosstalk compensation signal.
7. The computer-implemented method of any one of claims 3 to 6 wherein the device is a computer-based audiometer having channels that are not fully isolated.
8. A computer program product comprising a computer readable memory storing computer executable instructions thereon that when executed by a computer with channels that are not fully isolated performs the method steps of claims 3, 4, 5, and 6.
9. A computer with channels that are not fully isolated that performs the computer-implemented method of claims 3, 4, 5, and 6 for reducing undesired crosstalk signals on an inactive channel of the computer.
10.A computer-based audiometer with channels that are not fully isolated that performs the computer-implemented method of claims 3, 4, 5, and 6 for reducing undesired crosstalk signals on an inactive channel of the computer-based audiometer.
Priority Applications (4)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CA2908891A CA2908891A1 (en) | 2015-10-15 | 2015-10-15 | A computer-implemented method for reducing crosstalk in a computer-based audiometer |
| US15/768,355 US10506341B2 (en) | 2015-10-15 | 2016-10-06 | Computer-implemented method for reducing crosstalk in a computer-based audiometer |
| CA3000433A CA3000433C (en) | 2015-10-15 | 2016-10-06 | A computer-implemented method for reducing crosstalk in a computer-based audiometer |
| PCT/CA2016/000249 WO2017063069A1 (en) | 2015-10-15 | 2016-10-06 | A computer-implemented method for reducing crosstalk in a computer-based audiometer |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CA2908891A CA2908891A1 (en) | 2015-10-15 | 2015-10-15 | A computer-implemented method for reducing crosstalk in a computer-based audiometer |
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| Publication Number | Publication Date |
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| CA2908891A1 true CA2908891A1 (en) | 2017-04-15 |
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| CA2908891A Abandoned CA2908891A1 (en) | 2015-10-15 | 2015-10-15 | A computer-implemented method for reducing crosstalk in a computer-based audiometer |
| CA3000433A Expired - Fee Related CA3000433C (en) | 2015-10-15 | 2016-10-06 | A computer-implemented method for reducing crosstalk in a computer-based audiometer |
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| CA3000433A Expired - Fee Related CA3000433C (en) | 2015-10-15 | 2016-10-06 | A computer-implemented method for reducing crosstalk in a computer-based audiometer |
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| Country | Link |
|---|---|
| US (1) | US10506341B2 (en) |
| CA (2) | CA2908891A1 (en) |
| WO (1) | WO2017063069A1 (en) |
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| GB2596169B (en) * | 2020-02-11 | 2022-04-27 | Tymphany Acoustic Tech Ltd | A method and an audio processing unit for detecting a tone |
| TWI764296B (en) * | 2020-09-25 | 2022-05-11 | 財團法人工業技術研究院 | Electronic device and method for signal interference compensation |
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| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US3714595A (en) * | 1971-03-25 | 1973-01-30 | Warwick Electronics Inc | Demodulator using a phase locked loop |
| US20100027799A1 (en) * | 2008-07-31 | 2010-02-04 | Sony Ericsson Mobile Communications Ab | Asymmetrical delay audio crosstalk cancellation systems, methods and electronic devices including the same |
| JP5691130B2 (en) * | 2009-03-11 | 2015-04-01 | ヤマハ株式会社 | Apparatus, method, program, and system for canceling crosstalk when performing sound reproduction with a plurality of speakers arranged to surround a listener |
| US20130156238A1 (en) * | 2011-11-28 | 2013-06-20 | Sony Mobile Communications Ab | Adaptive crosstalk rejection |
| US9339216B2 (en) * | 2012-04-13 | 2016-05-17 | The United States Of America As Represented By The Department Of Veterans Affairs | Systems and methods for the screening and monitoring of inner ear function |
-
2015
- 2015-10-15 CA CA2908891A patent/CA2908891A1/en not_active Abandoned
-
2016
- 2016-10-06 CA CA3000433A patent/CA3000433C/en not_active Expired - Fee Related
- 2016-10-06 US US15/768,355 patent/US10506341B2/en not_active Expired - Fee Related
- 2016-10-06 WO PCT/CA2016/000249 patent/WO2017063069A1/en active Application Filing
Also Published As
| Publication number | Publication date |
|---|---|
| US10506341B2 (en) | 2019-12-10 |
| CA3000433C (en) | 2020-06-30 |
| WO2017063069A1 (en) | 2017-04-20 |
| US20180317008A1 (en) | 2018-11-01 |
| CA3000433A1 (en) | 2017-04-20 |
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